Networked computer telephony system driven by web-based applications

ABSTRACT

A networked telephony system and method allow users to deploy on the Internet computer telephony applications associated with designated telephone numbers. The telephony application is easily created by a user in XML (Extensible Markup Language) with predefined telephony XML tags and easily deployed on a website. The telephony XML tags include those for call control and media manipulation. A call to anyone of these designated telephone numbers may originate from anyone of the networked telephone system such as the PSTN (Public Switched Telephone System), a wireless network, or the Internet. The call is received by an application gateway center (AGC) installed on the Internet. Analogous to a web browser, the AGC provides facility for retrieving the associated XML application from its website and processing the call accordingly. The architecture and design of the system allow for reliability, high quality-of-service, easy scalability and the ability to incorporate additional telephony hardware and software and protocols.

CROSS-REFERENCE TO RELATED APPLICATION

This application is a divisional of application Ser. No. 09/675,497,filed Sep. 29, 2000, which application is incorporated herein in itsentirety by this reference.

FIELD OF THE INVENTION

The present invention relates to telecommunication, and moreparticularly to a networked computer telephony system including theInternet and the Public Switched Telephone System and driven byXML-based telephony applications distributed on the Internet.

BACKGROUND OF THE INVENTION

Two major telecommunication networks have evolved worldwide. The firstis a network of telephone systems in the form of the Public SwitchedTelephone System (PSTN). This network was initially designed to carryvoice communication, but later also adapted to transport data. Thesecond is a network of computer systems in the form of the Internet. TheInternet has been designed to carry data but also increasingly beingused to transport voice and multimedia information. Computersimplementing telephony applications have been integrated into both ofthese telecommunication networks to provide enhanced communicationservices. For example on the PSTN, computer telephony integration hasprovided more functions and control to the POTS (Plain Old TelephoneServices). On the Internet, computers are themselves terminal equipmentfor voice communication as well as serving as intelligent routers andcontrollers for a host of terminal equipment.

FIG. 1A illustrates a typical configuration of a conventional computertelephony server operating with a Public Switched Telephone Network(PSTN) and/or the Internet. Telephone service is traditionally carriedby the PSTN. The PSTN 10 includes a network of interconnected localexchanges or switches 12. Around each exchange is provisioned a clusterof telephone lines to which telephones, modems, and facsimile machinesmay be attached. Other private exchanges such as Private Brach Exchange(PBX) 20 may also be connected to the PSTN to form a public/privatetelephone network. Voice or data is carried from a source node to adestination node on the network by establishing a circuit path along thePSTN effected by appropriately switching the interconnecting exchanges.The point-to-point transmission is therefore circuit-switched,synchronous and using a dedicated channel of fixed bandwidth (64 kbs).With the introduction of digital networks, the exchanges have mostlybeen upgraded to handle digital, time-division multiplexed trunk trafficbetween the exchanges. External digital communication systems typicallycommunicate with the PSTN by interfacing with an exchange such as 12. Acommon digital interface at the exchange is PRI (Primary RateInterface), which is part of an ISDN (Integrated Services DigitalNetwork) and is usually provided by a T1 or E1 trunk line. Depending onthe bandwidth requirement of the external system, the interface with anexchange may require from one to a multiple of PRI connections.

The Internet 30 is a worldwide interconnection of IP (Internet Protocol)networks, with interconnecting computers communicating with each otherusing TCP/IP (Transmission Control Protocol/Internet Protocol). Some ofthe computers may also be interconnected by a private segment of the IPnetwork with restricted access. On an IP network, data from a sourcenode is cast into a number of packets that may individually betransported via multiple paths on the network to be reassembled at adestination node. The transmission on the IP network is packet-switchedand asynchronous.

On an IP network, voice or multimedia information can also be digitizedas data and transported over the network using the Internet Protocol(IP). In that case, it is generally referred to as VoIP or(Voice-over-IP). The H.323 standard promulgated by the ITU(International Telecommunication Union) aims to ensure VoIPinteroperability. It provides a specification for communication ofmultimedia such as voice, data and video between terminal equipment overIP networks. The terminal equipment communicating on the Internetincludes personal computers with telephony capabilities 40, VoIP phones42 that can connect to the Internet directly, and other networkedtelephony appliances.

In recent years, the World Wide Web (WWW) has become a universalplatform for information dissemination on the Internet. Web applications44 in general and web pages in particular are written in HTML (HyperTextMarkup Language) and are hosted by web servers 46 on the Internet. Eachweb page can be called up by its URL (Uniform Resource Locator), whichis its IP address on the Internet. These web pages may be requested andprocessed by a web browser running on a computer connected to theInternet. The web browser retrieves the web page under HTTP (HyperTextTransfer Protocol) and parses the HTML codes on the web page to executeit. Typically, the execution of HTML codes on a web page results inrendering it into a display page on the browser or client computer. Inother instances, it may result in the execution of some backendfunctions on the client and/or server computers. One reason for thewidespread acceptance of the WWW is the relative ease with which webapplications can be created and deployed, and the existence ofstandardized web browsers. HTML, with its tag-coding scheme, is now wellknown to everyone from the professional developer to the savvy end user.More recently, XML (Extensible Markup Language) has been introduced toextend HTML with enhanced features including customizable tags thatallow for more structural specification of data.

Telephony or Computer Telephony Integration (CTI) involves using acomputer to control and manage a phone or a telephone system. Whenapplied to a phone or a terminal equipment, CTI provides added featuresto an end user's phone. When applied to a telephone system whether aspart of the PSTN or part of an IP telephony network system, CTI isusually implemented with a CT (Computer Telephony) server, such as CTserver 50. Such a server executes telephony applications that canprovide custom services such as interactive voice response, customerservice or help desk for an organization. The CT server 50 can beconfigured to interface via a PSTN interface 52 with an exchange 12 toreceive and process calls pertaining to a predefined set of telephonenumbers on the PSTN. Similarly, it can also be configured to interfacevia an IP network interface 54 with the Internet to receive and processcalls pertaining to a predefined set of telephone numbers or IPaddresses. The CT server 50 is usually a computer operating under UNIXor Microsoft Windows NT and is running installed customized applicationsoftware 56 for the various voice applications. The CT server provides aset of APIs 58 (Application Program Interfaces) which are procedures,protocols and tools for building software applications. These APIs aregenerally proprietary and specific to the individual hardwaremanufacturers. Developing an application on an existing CT server wouldinvolve a highly specialized application developer undertaking a fairlycomplex task of coding the application in C++ or JAVA programminglanguage and employing and invoking the APIs specific to the hardware.

U.S. Pat. No. 6,011,844 discloses a distributed call center system inwhich a business call center running a custom interactive voice responseapplication is essentially replicated in a number of local points ofpresence to reduce communication cost when connecting a local customer.

FIG. 1B illustrates a Point-Of-Presence call center management systemdisclosed in U.S. Pat. No. 6,011,844. The system is designed to minimizelong distance toll call when a customer 70 is calling a business callcenter 60. The business call center typically runs a customizedinteractive voice response application 66 that implements a completebusiness solution to answer, service, queue and route inbound customercalls. The customer 70 at a local exchange 72 will in general be callinglong distance to the business call center 60 that is local to a remoteexchange 74. When the customer requests to speak to a live agent 68 atthe business call center, his or her call is queued until an agent isavailable. Thus, during the long distance call, apart from interactingwith the interactive voice responses, a substantial portion of timecould be incurred while waiting to speak to an agent. To reduce the longdistance connection time, the POP call center management system deploysa number of POP call centers 80 across the Public Switched TelephoneNetwork (PSTN) 10 so that a customer's call at a local exchange 72 isintercepted at a local POP call center 80. Each POP call centeressentially serves as a local-presence business call center exceptwithout the live agent. This is accomplished by having each POP callcenter executing the application such as 66′, 66″ locally. The localapplications 66′, 66″ can be full replicas of the application 66residing at the business call center or they can be partial ones withsome of the resources such as voice prompts, menus, etc., being accesseddynamically from the application 66 as needed. The application 66 thatresides at the business call center is accessible by the POP callcenters via an interconnecting virtual private network 90. Optionally,HTML or XML may be used when the POP call center accesses convenientlypackaged units of information or applications from the business callcenter across the call center virtual private network 90. Thus, with theexception of speaking to a live agent, the customer's call is basicallyhandled at a POP call center local to the customer. When the customerrequests to speak to a live agent, a queue is set up at the businesscall center until an agent becomes available. Only then will the POPcall center convert the customer's local call to a long distance call tothe business call center. The voice traffic for the interactive voiceresponse portion is carried between the local exchange 72 and a POP callcenter 80. The voice traffic between the customer 70 and a live agent 68is carried via a long distance portion 76 of PSTN, or in other disclosedembodiments, over the call center virtual private network 90 or theInternet 30.

Prior computer telephony systems have infrastructures that do not alloweasy development and deployment of telephony applications. The systemillustrated in FIG. 1A requires the telephony application to be hostedin a call center type of telephony server and requires specializedknowledge of the telephony hardware to develop telephony applications.The same is true for the system illustrated in FIG. 1B with thevariation that the call center is effectively replicated at variouslocal points of presence on the global telephone network.

SUMMARY AND OBJECTS OF THE INVENTION

It is therefore a general object of the invention to provide a computertelephony system that allows easy development and deployment oftelephony applications.

It is another general object of the invention to provide aninfrastructure in which a large number of developers and end users caneasily create and deploy custom telephony applications for controllingand managing telephone calls on the PSTN and the Internet.

It is another object of the invention to have a computer telephonysystem that provides an application development and deploymentenvironment similar to that for HTML applications and the World WideWeb.

It is another object of the invention to provide a low cost routing oftelephone calls among the interconnected PSTN and Internet.

It is yet another object of the invention to provide a telecommunicationnetwork with improved quality of service.

These and other objects of the invention are accomplished, briefly, byproviding a networked computer telephony system that includes creatingtelephony applications in XML scripts that include telephony-specificXML tags specifying how a telephone call to a specified call number isto be processed. The XML scripts associated with each specific callnumber are posted on web servers on the Internet. A telephone call tothe specified call number is routed through the Internet to anapplication gateway center. The application gateway center retrieves theassociated XML scripts and executing the scripts to process the call.

In a preferred embodiment, a plurality of application gateway centersare installed on the Internet to provide for reliability, scalabilityand high quality of service.

In a preferred embodiment, the application gateway center includes acache server for caching data exchanged between the application centerand the Internet.

In a preferred embodiment, the application gateway center manipulatesmedia in a predefined native format; and the application gateway centerincludes a media conversion proxy server for converting between saidpredefined format native to the application gateway center and othermedia formats outside of the application gateway center.

According to another aspect of the invention, a method of processing atelephone call to a specified call number includes providing anExtensible Markup Language (XML) document associated with the specifiedcall number, said XML document constituting a telephony application andincluding telephony-specific XML tags instructing how a telephone callto the specified call number is to be processed; posting said XMLdocument to a specified location on the Internet; providing a directoryfor locating said XML document by the specified call number; receivingsaid telephone call on the Internet; retrieving said XML document at thespecified location looked up from said directory with the specified callnumber; and processing said telephone call according to said XMLdocument.

According to another aspect of the invention, in order to provide highquality of service, the networked computer telephony system furtherincludes a plurality of network traffic monitors. Each monitor isassociated with an individual application gateway center forperiodically monitoring network traffic statistics regarding a responsetime of a specific XML document being requested by a specificapplication gateway center. A network monitoring server dynamicallyanalyzes said network statistics collected from said plurality ofnetwork traffic monitors into a prioritized list of XML documentsrelative to application gateway centers having the fastest accessthereto. The prioritized list is used for directing a telephone call toa specific call number to the application gateway with the fastestaccess to the XML document associated with the specific call number.

Additional objects, features and advantages of the present inventionwill be understood from the following description of its preferredembodiments, which description should be taken in conjunction with theaccompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1A illustrates a typical configuration of a conventional computertelephony server operating with a Public Switched Telephone Network(PSTN) and/or the Internet.

FIG. 1B illustrates a Point-Of-Presence call center management systemdisclosed in U.S. Pat. No. 6,011,844.

FIG. 2 illustrates an Application Gateway Center (vAGC) for processingtelephony applications on the Internet and the PSTN, according to ageneral scheme of the present invention.

FIG. 3 is a flow diagram illustrating the setup for provisioning andprocessing voice applications according to a general embodiment of thepresent invention.

FIG. 4A illustrates a preferred configuration of the inventive systemwith respect to the Internet and the PSTN.

FIG. 4B is a flow diagram illustrating an exemplary call routing andprocessing in the preferred configuration shown in FIG. 4A.

FIG. 5 illustrates an alternative preferred configuration of theinventive system with respect to the Internet and the PSTN.

FIG. 6 is a block diagram illustrating the components of the ApplicationGateway Center.

FIG. 7 is a block diagram illustrating schematically the components ofthe media conversion proxy server.

FIG. 8 is a detailed block diagram of the Application Gateway Server,which is the main component of the Application Gateway Center.

FIG. 9 is a system block diagram of a network traffic monitoring systemoperating in cooperation with the Distributed Application TelephonyNetwork System of the present invention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

As mentioned in an earlier section, the Internet is a worldwide networkof IP networks communicating under TCP/IP. Specifically, voice and othermultimedia information are transported on the Internet under the VoIP(Voice-over-IP) protocol, and particularly under the H.323 standard thathas been put forward for interoperability.

FIG. 2 shows a typical environment including the Internet and the PSTNin which the present invention is practiced. The Internet 30 acts as aVoIP network for communication between terminal equipments, such aspersonal computers (PC) 40 with telephony capabilities and/or dedicatedVoIP phones 42 connectable directly to the Internet. Each terminalequipment on the Internet has an IP address that may also be associatedwith a predefined call number so that one terminal equipment may callanother one by its IP address or equivalently by its call number. Alsodeployed on the Internet are HTML applications such as an application 44hosted on a web server 46 that may also interact with other clients andservers 48 on the Internet.

On the other hand, the PSTN 10 is a network of exchanges. Each exchangeis provisioned with a plurality of telephone lines or nodes havingdesignated call numbers. Two PSTN nodes are connectable by switching theintervening exchanges to form a circuit.

The PSTN and the Internet are interconnected by means of access serverssuch as an access server 14. This enables communication between a PSTNnode and an Internet node. A telephonic call transported between twonetwork nodes comprises a signaling portion for setting up and tearingdown the call and a media portion for carrying the voice or multimediadata. The access server 14 essentially converts both of these portionsto an appropriate format across the interface between the two types ofnetworks. On the PSTN side the digital interface is PRI and on theInternet side the interface is VoIP. A wireless or mobile telephonenetwork (not shown) may similarly be considered as an extension of thePSTN. It is typically connected to the PSTN via a suitable interfaceimplemented by a gateway.

FIG. 2 illustrates an Application Gateway Center (vAGC) for processingtelephony applications on the Internet and the PSTN, according to ageneral scheme of the present invention. The Application Gateway Center(vAGC) 100 is a call-processing center on the Internet 30 forintercepting and processing calls to any one of a set of designatedtelephone call numbers. The calls may originate or terminate on anynumber of interconnected telecommunication networks including theInternet 30, the PSTN 10, and others (not shown) such as wirelessnetworks. The vAGC 100 processes each call according to the telephonyapplication (vAPP) associated with the called number. A plurality ofthese associated telephony application, vAPPs, such as 110, . . . ,110′, are deployed on the Internet in the form of XML applications.These XML applications, denoted more specifically as (vXML)applications, are coded in XML scripts that also contain customtelephony XML tags. The vXML scripts allow complete telephonyapplications to be coded.

The set of designated call numbers handled by the vAGC 100 areregistered in a directory, such as DIR0. When a call to one of thedesignated call numbers is made from the PSTN, it is switched to theaccess server 12 and a lookup of the directory DIR0 allows the call tobe routed to vAGC 100 for processing. Similarly, if the call originatesfrom one of the terminal equipment on the Internet, a directory lookupof DIR0 provides the pointer for routing the call to the vAGC 100.

The plurality of telephony applications vAPP 110, . . . , 110′, eachassociated with at least one designated call number is accessible by thevAGC from the Internet. Each application is coded in vXML and is beinghosted as a webpage on a web server on the Internet. A directory DIR1provides the network address of the various applications. When the vAGC100 received a call, it uses the call number (or dialed number DN) tolook up DIR1 for the IP address of the vAPP associated with the DN. ThevAGC 100 retrieves the vXML webpage and executes the call according tothe vXML scripts.

FIG. 3 is a flow diagram illustrating the setup for provisioning andprocessing voice applications according to a general embodiment of thepresent invention. Provisioning of a designated call number with itsassociated vAPP is described in steps 130, 132 and 134.

-   Step 130: For a given call number DN, create an associated telephony    application, vAPP in vXML, and deploy it on the Internet with a    specific IP address or URL.-   Step 132: Provide any media, files and web applications that are    requested or act on by vAPP.-   Step 134: Update the directory DIR1 so that the address of vAPP can    be obtained by querying with its associated call number DN.

Call processing by vAGC 100 is described in steps 140, 142, 144 and 146.

-   Step 140: vAGC receives a call with DN routed thereto.-   Step 142: vAGC uses DN to look up DIR1 for the address of the    webpage for vAPP.-   Step 144: vAGC requests and retrieves the webpage containing vXML    scripts for vAPP.-   Step 146: vAGC processes the call according to the retrieved vXML    scripts for vAPP.

FIG. 4A illustrates a preferred configuration of the inventive systemwith respect to the Internet and the PSTN. The configuration is similarto that shown in FIG. 2 except there are a plurality of ApplicationGateway Centers (vAGCs) 100, 100′, . . . , 100″ deployed on the Internet30. This will provide redundancy, capacity and load-balancing forexecuting the plurality of telephony applications vAPP 110, . . . , 110″being hosted by webservers 112, . . . , 112′ on the Internet. In orderto provide local access to the Internet 30 from anywhere on the PSTN 10,individual Local Exchange Carriers (LECs) covering the PSTN are providedwith an access server (AS). Each access server communicates on the onehand with an exchange of the LEC via the PRI interface and on the otherhand with the Internet via the H.323 VoIP standard. In this way, a callmade at most nodes on the PSTN can be routed to the Internet withoutincurring a toll call outside an LEC domain.

FIG. 4B is a flow diagram illustrating an exemplary call routing andprocessing in the preferred configuration shown in FIG. 4A. The numeralin parenthesis denotes the route taken. A new call originates from atelephone line 11 on a local exchange. Since the call is made to adialed number (DN) registered as one of the numbers handled by the vAGC,it is routed to a vAGC such as vAGC 100 after a lookup from DIR0. ThevAGC 100 initiates a new session for the call and looks up DIR1 for thenet address of the telephony application vAPP 110 associated with theDN. The vAGC 100 retrieves vAPP 110 and proceeds to process the vXMLscripts of vAPP 110. In one example, the vXML scripts dictate that thenew call is to be effectively routed back to the PSTN to a telephone 13on another local exchange. In another example, the vXML scripts dictatethat the call is to be effectively routed to a VoIP phone 15 on theInternet. In practice, when connecting between two nodes, the vAGCcreates separate sessions for the two nodes and then bridges orconferences them together. This general scheme allows conferencingbetween multiple parties. In yet another example, the vXML scriptsallows the call to interact with other HTML applications or otherbackend databases to perform on-line transactions.

Thus, the present system allows very powerful yet simple telephonyapplications to be built and deployed on the Internet. The following aresome examples of the vAPP telephony applications contemplated. A“follow-me-find-me” application sequentially calls a series of telephonenumbers as specified by a user until one of the numbers answers and thenconnects the call. Otherwise, it does something else such as takes amessage or sends e-mail or sends the call to a call center, etc. Inanother example, a Telephonic Polling application looks up from adatabase the telephone numbers of a population to be polled. It thencalls the numbers in parallel, limited only by the maximum number ofconcurrent sessions supported, and plays a series of interactive voiceprompts/messages in response to the called party's responses and recordsthe result in a database, etc. In another example, a Help Deskapplication plays a series of interactive voice prompts/messages inresponse to the called party's responses and possibly connects the callto a live agent as one option, etc. In yet another example, a Stock orBank Transactions application plays a series of interactive voiceprompts/messages in response to the called party's responses andconducts appropriate transactions with a backend database or webapplication, etc.

FIG. 5 illustrates an alternative preferred configuration of theinventive system with respect to the Internet 30 and the PSTN 10. Thearrangement is similar to that of FIG. 4A except at individual LECs, theApplication Gateway Centers vAGC 100, 100′, . . . , 100″ arerespectively co-located with the local access servers AS 14″, 14′, . . ., 14. This configuration provides higher quality-of-service (QoS) at theexpense of repeating the vAGC at every LEC.

FIG. 6 is a block diagram illustrating the components of the ApplicationGateway Center. The Application Gateway Center vAGC 100 may beconsidered to be a facility hosting a cluster of servers for the purposeof receiving calls and running the associated telephony applications,vAPPs, reliably and efficiently. In the preferred embodiment, the vAGC100 comprises two IP network segments. An Internet network segment 130connects the vAGC 100 to the Internet. A local IP network segment 140allows direct communication between an application gateway server 200, acache server 310 and a media conversion proxy server 320. The cacheserver 310 and the media conversion proxy server 320 are also connecteddirectly to the Internet via the Internet network segment 130. Toincrease performance and reliability, multiple servers of each type areinstalled in the vAGC 100.

The application gateway server 200 exchanges data with the Internetindirectly through the cache server 310 and possibly the mediaconversion proxy server 320. As will be described in more detail later,upon receiving a call, the AGS 200 retrieves the associated vAPP from awebsite and proceeds to execute the vXML scripts of the vAPP. During thecourse of executing the vXML scripts, associated media and/or files mayalso be retrieved from various sites as part of the vAPP suite.

In the preferred embodiment, in order to increase performance, the vXMLscripts, media and files that are retrieved into the vAGC are cached bythe cache server 310. They are requested by the AGS through the cacheserver 310. If a cached copy of the requested data exists in the cacheserver, it is delivered directly to the AGS. If not, the cache serverretrieves the data, caches it and delivers the data to the AGS tofulfill the request.

In the preferred embodiment, in order to simplify the design of the AGSand to improve the performance and scalability of it, the AGS isdesigned to handle only one native media format. For example, onesuitable format for audio is G.711 or GSM. Media that come in differentformat are handed over to the media conversion proxy server 320, whichcoverts the media to the native format of the AGS 200.

FIG. 7 is a block diagram illustrating schematically the components ofthe media conversion proxy server. The media conversion proxy servercomprises a text-to-speech module 322, a speech-to-text module 324, anaudio conversion module 326 and a protocol conversion module 328. Themodular design allows for other “plug-ins” as the need arises. Thetext-to-speech module 322 is used for converting text to synthesizedspeech. For example, this is useful for reading back e-mail messages.The speech-to-text module 324 is used for converting speech to text.This is useful in speech recognition applications involving respondingto a user's voice response. The audio conversion module 326 convertsbetween a supported set of audio formats, Such as G.711, G.723, CDaudio, MP3, etc. The protocol conversion module 328 allows conversionsbetween protocols such as IMAP (Internet Message Access Protocol) andSMTP (Simple Mail Transfer Protocol).

Application Gateway Server

FIG. 8 is a detailed block diagram of the Application Gateway Server,which is the main component of the Application Gateway Center. TheApplication Gateway Server (AGS) 200 is responsible for acceptingincoming calls, retrieving the vAPP associated with the dialed numberand executing the vXML scripts of the vAPP. Each incoming call istreated as a separate session and the AGS is responsible for processingall user events and system actions that occur in multiple simultaneoussessions. The AGS is also responsible for all call routing in allsessions.

In the preferred embodiment, the AGS 200 is a set of software modulesrunning on a Windows NT or Unix server. For example, the AGS isimplemented as a Windows NT machine on a card, and multiple cards areinstalled on a caged backplane to form a highly scalable system.

The AGS 200 comprises four main software modules: a session manager 210;an I/O abstraction layer 220; a computer telephony (CT) abstractionlayer 230; and a telephony scripting language parser 240. The telephonyscripting language parser 240 further comprises a telephony XML or vXMLparser 242 and a generic XML parser 244. In addition, a streaminginterface 250 provides a direct streaming path for media data betweenthe I/O abstraction layer 220 and the CT abstraction layer. Each ofthese modules is designed to be a separate DLL (Dynamically LinkedLibrary) and perform a specific task. In the preferred embodiment, theAGS is a console only application with no user interface for any ofthese modules. Several of these modules incorporate commercial, thirdparty software components in performing their tasks. These componentswill be discussed along with the appropriate modules.

The session manager 210 is the centerpiece of the AGS 200. It isresponsible for creating new sessions, deleting terminated sessions,routing all actions and events to the appropriate modules andmaintaining modularity between each session. It responds to I/O and vXMLgoto requests, and other additional events. In one embodiment, itemploys commercially available software libraries containing thread andstring classes from PWLib, a product of Equivalence Pty Ltd, Erina, NewSouth Wales, Australia.

The session manager interfaces to the external of the AGS via the I/Oabstraction layer 220 and the CT abstraction layer 230. It accesses theI/O and CT layers as a set of classes and member functions that areindividual DLLs. The Session Manager 210 runs as a single-threadedprocessor of actions and event.

FIG. 8 also illustrates the manner in which the modules of the AGS mustcommunicate with each other. The session manager communicates to boththe I/O abstraction layer and the CT abstraction layer throughtraditional DLL entry points with C/C++ parameter passing. The I/Oabstraction layer and the CT abstraction layer communicate through astreaming interface. The session manager and the telephony scriptinglanguage parser communicate through DLL entry points using microXML. Thesession manager 210 behaves like a virtual machine with its own set of“OpCodes”. MicroXML is the parsed vXML scripts interpreted into theseOpCodes, and will be described in more detail later.

A session begins with the reception of an asynchronous event from the CTabstraction module 230 signaling an incoming call. The Session Managerthen creates a session for this call by accessing a database (e.g. DIR1of FIG. 4A) keyed on the session's DNS and ANI information, whichreturns an initial vXML script. The telephony scripting language parser240 is a separate DLL invoked through short microXML event scripts. Itreturns a microXML action script. A cycle of actions and events beginswith the transmission of this script to the telephony scripting languageparser 240 for processing. The telephony scripting language parser 240responds to this event by returning a simple vXML script of its owncontaining I/O and CT action requests collected from the parsing of thescript. The Session Manager now processes these action requests and thenreturns to parsing until the end of the session.

Each session is assigned a unique session identification, SID (sessionID). For example, in the Microsoft Win32 platform, the SID isconveniently implemented by the creation of 128 bit globally unique Ids(GUIDs.

In the preferred embodiment, the session manager 210 is accessed orinvoked via a number of interface points of its DLL as described inTABLE 1.

TABLE 1 Session Manager's Interface Points SESSION MANAGER InterfacePoints DESCRIPTION VXESessionManager( ) VXESessionManager constructorfunction. It creates and starts up an instance of an AGS SessionManager. ~VXESessionManager( ) VXESessionManager destructor function. Itshuts down and deletes an instance of an AGS Session Manager.AddEvent(VXEEvent&) Member function to submit an event to a SessionManager's event queue. It receives a record of the incoming event andoutputs TRUE if submission is successful, FALSE, otherwise. GetSessions() Provides a count of active sessions.

The I/O abstraction layer 220 performs all input and output operationsfor the AGS 200. Essentially, it renders transparent to the internal ofthe AGS the variety of I/O formats and protocols that might be encounterexternally. To the session manager 210, most HTTP, FTP, File, andmemory-mapped I/O requests are reduced to four commands: open, close,read, and write. This allows access to a stream from any of thesesources with the same procedure calls once the stream is open. In oneembodiment, it incorporates available commercial software libraries,such as WinInet from Microsoft Corporation, Seattle; Wash., U.S.A andPWLib from Equivalence Pty Ltd. WinInet is a windows-specific DLL thatallows the I/O abstraction layer to communicate to outside sources usingHTTP and FTP. PWLib also used by the session manager 210 containsstrings and threads classes.

In the preferred embodiment, the I/O abstraction layer 220 is accessedor invoked via a number of interface points of its DLL as described inTABLE 2. A single thread per active stream is created by instantiating aVXEIOStream when accessed by the session manager 210. If the stream isFTP or HTTP-based, then the user will need to provide the appropriatelogin data, submission method, and CGI variables. Next, the user callsthe Open method and then uses the Read and Write methods to operate uponthe stream until closing it with the Close method. At this point, thisinstance of the VXEIOStream is available for use on another streamsource or it can be deleted.

TABLE 2 I/O Abstraction Layer's Interface Points I/O ABSTRACTION LAYERInterface Points DESCRIPTION VXEIOStream( ) VXEIOStream constructorfunction. It creates a new instance of a VXEIOStream. ~VXEIOStream( )VXEIOStream destructor function. It shuts down stream and releasesassociated memory open/openAsynchronous(char* Member function to open anI/O stream either name, StreamType streamtype, synchronously orasynchronously. It has inputs: OpenMode mode) pathname, type of stream(HTTP, FTP, File, or Memory), and opening mode (Read/Write); and output:TRUE/FALSE for success/failure in synchronous mode, corresponding eventasynchronously. close( ) Member function to close an open stream. Itoutputs: TRUE/FALSE for success/failure. read/readAsynchronous(void*Member function to read from an open stream either buffer, int count)synchronously or asynchronously. It has inputs: Pointer to buffer intowhich to write data, byte count to read from stream. It has outputs:Number of bytes read synchronously, corresponding event asynchronouslywrite/writeAsynchronous(void* Member function to write to an open streameither buffer, int count) synchronously or asynchronously. It hasinputs: Pointer to buffer from which to write data, byte count to writeto stream. It has outputs: Number of bytes written synchronously,corresponding event asynchronously. GetPos( ) Member function to returnposition within a stream. SetSubmitMethod(SubmitMethod Member functionto set CGI submission method for method) an HTTP stream before openingit. It has inputs: Submission method, either GET or PUT. AddCGIVariable(VXEVariable& v) Member function to add a CGI variable for submission toan HTTP stream before opening it. It has inputs: Variable name/valuepair contained in a VXEVariable class. It has outputs: TRUE/FALSE forsuccess/failure. SetFTPLogin(PString& name, Member function to set FTPlogin information for an Pstring& passwd) FTP stream before opening it.It has inputs: FTP user name and password.

The computer telephony (CT) abstraction layer 230 is a thin abstractionlayer that makes it possible for the AGS 200 to communicate with severalcomputer telephony devices and/or protocols. In one direction, the CTabstraction layer receives requests for computer telephony actions fromthe session manager 210 and translates those requests to a CT module. Inthe other direction the CT abstraction layer receives user eventsdirected to that CT module and relates them back to the session manager.In the preferred embodiment, the CT modules include a H.232 stack forhandling VoIP signals, a SIP (Session Interface Protocol), a MGCP (MediaGateway Control Protocol) as well as other CT modules such as DialogicCT modules. Since several CT modules can be placed below the CTabstraction layer and the CT abstraction will talk to all of the CTmodules, the modular design allows the AGS to communicate with a newcomputer telephony device or protocol simply with the addition of a newCT module.

The CT abstraction layer 230 will preferably make use of PWLib'splatform-independent thread class. The CT Abstraction layer isinstantiated by the Session Manager 210. It then seeks out a vXMLconfiguration file that contains information on the number and type oftelephony boards in its system. The member functions represent genericfunctionality that should be supportable across a wide variety oftelephony hardware. The motivation for this abstraction layer is to makethe AGS 200 both platform and protocol independent.

In the preferred embodiment, the Session Manager 210, XML Parser 240,and CT Abstraction layer 230 cooperate via the following protocol.First, the telephony scripting language parser 240 locates a vXMLelement which requires a telephony task. Next, the telephony scriptinglanguage parser sends this task to the Session Manager in a microXMLaction string. The Session Manager then parses the microXML actionstring and determines the appropriate call to the CT abstraction layeralong with its associated parameters. The Session Manager now calls theCT abstraction layer asynchronously and the CT abstraction layer returnsan event signaling the completion of the CT task and the Session Managerresumes parsing.

In the preferred embodiment, the CT abstraction layer 230 is accessed orinvoked via a number of interface points of its DLL as described inTABLE 3.

TABLE 3 CT Abstraction Layer's Interface Points CT ABSTRACTION LAYERInterface Points DESCRIPTION VXECTAbstraction(VXESessionManager*)VXECTAbstraction constructor function. It has input: Associated SessionManager. It creates a new instance of a CT Abstraction.~VXECTAbstraction( ) VXECTAbstraction destructor function. It shuts downan instance of a Voxeo CT Abstraction and releases associated memoryGetVersion(PString& version) Member function to determine version. Ithas inputs: Reference to a string into which to copy versioninformation. It has outputs: Version information copied into parameter 1string GetProtocol(PString& protocol) Member function to determineactive telephony protocol. It has inputs: Reference to a string intowhich to copy protocol information. It has outputs: Protocol informationcopied into parameter 1 string. Answer(VXESession* pSession) Memberfunction to answer an incoming call. It has inputs: Session associatedwith incoming call. It has outputs: Asynchronous event indicatingsuccess/failure sent to Session Manager. Hangup(VXESession* pSession)Member function to hang up on an active call. It has inputs: Sessionassociated with active call. It has outputs: Asynchronous eventindicating success/failure sent to Session Manager. call(VXESession*pSession, Member function to make an outgoing call. It has VXECall*)inputs: Associated session, number to call. It has outputs: Asynchronousevent indicating success/failure sent to Session Manager.dial(VXESession* pSession, Pstring* Member function to dial a string ofdigits. It has number) inputs: Associated session, digits to dial. Ithas outputs: Asynchronous event indicating success/failure sent toSession Manager. Wink(VXESession* pSession) Member function to performwink function. It has inputs: Associated session. It has outputs:Asynchronous event indicating success/failure sent to Session Manager toan HTTP stream before opening it. Void conference(VXESession* Memberfunction to conference two active pSession1, VXESession* pSession2)sessions/calls. It has inputs: Two sessions to conference together. Ithas outputs: Asynchronous event indicating success/failure sent toSession Manager. Void flushDigitBuffer(VXESession* Member function toflush digit buffer. It has inputs: pSession) Associated session. It hasoutputs: Asynchronous event indicating success/failure sent to SessionManager. Void getDigits(VXESession* Member function to read digits fromdigit buffer. It pSession, int maxdigits, Pstring& has inputs:Associated session, maximum digits to termdigits, Pstring& outdigits)read, termination digits string, string for digits read. It has outputs:Asynchronous event indicating success/failure and digits read sent toSession Manager. PlayStream(VXESession* Member function to play audiofrom an open stream. pSession, VXEIOStream&, const It has inputs:Associated session, audio stream, and Pstring& termdigits) terminationdigits. It has outputs: Asynchronous event indicatingcompletion/termination sent to Session Manager. PlayDate(VXESession*pSession, Member function to play current date. It has inputs: constPString& date, const PString& Associated session, string containingdesired date, termdigits) termination digits string. It has outputs:Asynchronous event indicating completion/termination sent to SessionManager. PlayTime(VXESession* pSession, Member function to play currenttime. It has inputs: const PString& time, const PString& Associatedsession, string containing desired time, termdigits) termination digitsstring. It has outputs: Asynchronous event indicatingcompletion/termination sent to Session Manager. PlayMoney(VXESession*pSession, Member function to play a dollar value. It has const floatvalue, const PString& inputs: Associated session, value to play,termdigits) termination digits string. It has outputs: Asynchronousevent indicating completion/termination sent to Session Manager.PlayCharacters(VXESession* Member function to play a string ofcharacters. It pSession, const PString& string, has inputs: Associatedsession, string of characters const Pstring& termdigits) to play,termination digits. It has outputs: Asynchronous event indicatingcompletion/termination sent to Session Manager. PlayString(VXESession*pSession, Member function to pronounce a text message. It const PString&string, const has inputs: Associated session, string to pronounce,Pstring& termdigits) termination digits. It has outputs: Asynchronousevent indicating completion/termination sent to Session Manager.PlayNumber(VXESession* pSession, Member function to play a number. Ithas inputs: const PString& number, const Associated session, stringcontaining number to Pstring&termdigits) pronounce, termination digits.It has outputs: Asynchronous event indicating completion/terminationsent to Session Manager. PlayOrdinal(VXESession* pSession, Memberfunction to play an ordinal (1st, 2nd, 2rd, const PString& ordinal,const etc.). It has inputs: Associated session, ordinal, Pstring&termdigits) termination digits. It has outputs: Asynchronous eventindicating completion/termination sent to Session Manager.RecordStream(VXESession* Member function record to an open VXEIOStream.pSession, XEIOStream& stream) It has inputs: Associated session, targetstream. It has outputs: Asynchronous event indicating success/failuresent to Session Manager. SendFAX(VXESession* pSession, Member functionto send a FAX. It has inputs: VXEIOStream& file) Associated session,VXEIOStream containing data to FAX. It has outputs: Asynchronous eventindicating success/failure sent to Session Manager.ReceiveFAX(VXESession* Member function to receive a FAX. It has inputs:pSession, VXEIOStream &file) Associated session, VXEIOStream to which toreceive FAX. It has outputs: Asynchronous event indicatingsuccess/failure sent to Session Manager.

The streaming interface 222 provides a direct streaming transfer betweenthe I/O abstraction layer 226 and the CT abstraction layer 230 whenmedia data, such as audio or other multimedia is involved. For example,the streaming interface facilitates the AGS to play audio from URL's andto record audio to URL's in a streaming manner. In the preferredembodiment, the interface is generic and passes the burden of buffermanagement to the CT module in use. This allows specific CT modules tobuffer information as appropriate for the corresponding telephonyhardware or protocol. The streaming interface is implemented through thereadAsynchronous and writeAsynchronous interface points in the I/Oabstraction layer.

The telephony scripting language parser 240 is responsible for parsingthe vXML scripts handed to it by the session manger 210. It in turninforms the session manager of the described actions coded in the vXMLscripts. The telephony scripting language parser is modular and canaccommodate additional parsers such as that for voiceXML and parsers forother telephony scripting language that may arise. In the presentpreferred embodiment, it comprises the vXML parser 242 and the genericXML parser 244

The generic XML parser 244 parses the vXML scripts, which areessentially XML scripts with embedded custom telephony tags, and putsthem in a format that the vXML parser 242 can expediently act on. In thepreferred embodiment, the generic XML parser 244 conveniently employsCueXML components available from CueSoft, Inc, Brighton, Colo., U.S.A.These components enable parsing of vXML documents into an object model,DOM (Document Object Model) listing the parsed objects in a hierarchicaltree structure. This allows the vXML parser 242, which in the preferredembodiment is a DLL written in Delphi 5.0, to “walk” through the tree ofobjects and interpret them into microXML codes that can be understood bythe session manager 210.

The vXML parser 242 behaves as follows: when called it will examine theincoming microXML and determine if there is a buffer of new vXML toparse, if such a buffer exists then the parser uses the generic XMLparser 244 to construct a new object model for this buffer, the sessionobject model is set to that model and the session state is cleared. ThevXML parser 242 begins parsing from the session state in the sessionobject model (an empty state implies the beginning of a document). Asthe parse traverses the document model the state is updated and eventsare generated. If these events are internal to the processor they arehandled (i.e. assigns update the session variables, blocks may causelooping to occur), if the events are not internal then they are bufferedfor return to the session manager. When an event needs to be reported tothe session manager the event buffer is processed so that variables arereplaced with their values, wildcards are properly expanded, etc. Thisnegates the need for any other module to maintain information aboutsession variables.

The vXML parser 242 is required to maintain state per session so thateach invocation of the vXML parser will continue where the previousinvocation within the same session ended. The maintenance of stateincludes preserving the DOM for the current instance of vXML, the nodein the DOM that the parser is currently examining, and any variablesthat are associated with the session.

In the preferred embodiment, the vXML parser 242 is accessed or invokedvia a number of interface points of its DLL as described in TABLE 4.

TABLE 4 vXML Parser Interface Points VXML PARSER Interface PointsDESCRIPTION Create Creates an instance of the vXML parser. It hasoutput: integer result code (negative numbers denote errors). DestroyDestroys an instance of the vXML parser. It has output: integer resultcode (negative numbers denote errors). Parse Performs the main tasks ofthe vXML parser (i.e. determines actions from vXML, and maintains state.It inputs: microXML string containing the sessionID. The microXML mayalso contain a buffer of vXML (described above) and a pointer toinstance data. It outputs: microXML string containing the action(s)generated by this invocation and possibly modification of the instancedata. Kill It has input: pointer to instance data. It has output:integer result code (negative numbers denote errors).

As mentioned earlier, microXML is a subset of simple vXML used forcommunication between the session manager 210 and the telephonyscripting language parser 240. MicroXML is the native codes of thevirtual machine of the session manager 210. In one direction, the vXMLparser 242 communicates with the session manger 210 in a synchronousmanner using microXML. In another other direction, user events may alsobe reported to the vXML parser via microXML. If a user event is reportedthe parser will find the appropriate event handler by first lookinglocally for a valid handler. If a handler is not found there then theparent node in the document model is examined for a valid handler. Thesearch continues in this manner until either a handler is found or thereis no parent to examine. If a handler is found then the parser sets thestate to the handler and begins parsing as described above. If a handleris not found then an error is returned via microXML.

In the preferred embodiment, MicroXML is composed of a limited number oftags, these tags do not have any attributes, and CDATA sections are notsupported. Table 5 shows examples of microXML tags:

TABLE 5 microXML Tags MicroXML TAG NAME ACT Action BUF Buffer DATInstance Data ERR Error EVT Event ETP Event Type EVL Event Value LBLLabel TYP Type P00 Parameter0 P00 Parameter99 SID Session ID

vXML is XML with additional custom tags for telephony applications.TABLE 6A-6D lists example tags useful for creating telephonyapplications. A user or developer need only code his or her telephonyapplication in these vXML tags and deploy the resulting scripts as awebpage on the Internet for the vAGS 200 to access.

TABLE 6A vXML General Tags vXML GENERAL TAG Examples DESCRIPTION <assignvar=“ttt” Assigns value “123” to variable named “ttt.   value=“123”/><clear var=“ttt”/> Clears variable named “ttt”. <clearDigits /> Clearsthe digit buffer. <getDigits This element reads input digits from the  var=“pager_msg” phone and places them into a variable defined  maxdigits=“9” within the element itself. In the example, the  termdigits=“#*” user would have 30 seconds to enter up to 9  includetermdigit=“TRUE|FALSE” digits on her phone, pausing no morethan 5   cleardigits=“TRUE|FALSE” seconds between digits, and ending thedigit   maxtime=“30s” input with either the # key or * key. This  ptime=“5s”/> element is designed for gathering PIN codes, Pagernumbers, and anything else that involves multiple digits coming from theuser. <block The block element is used to logically group   label=“here”other elements together, as well as providing   repeat=“?” a loopingstructure so that vXML elements   cleardigits=“TRUE|FALSE”> can berepeated a specific number of times Events (e.g., a menu that plays anaudio prompt four Elements times before timing out.) </block> <goto Thiselement will leap to another bank of value=“http://w.v.n/next.voxeo#block” vXML code, whether it be in thesame file or    submit=“all|*|x,y,z” another file.   method=“put|post”/> <return/> One example of Return is to implement<goto> calls as a call stack. <Return> would return from a <goto> call.<run This runs/launchs a vXML script at a URL or value=“http://w.v.n/next.voxeo|#block” URI in a new session, thencontinues to   submit=“all|*|x,y,z” process this session  method=“put|post”   newSessionID=“newID”/> <sendeventvalue=“msg_call_answered” This tag allows for one session to send a   sessionID=“sss”/> message to another session.

TABLE 6B vXML CAll Control Tags vXML CALL CONTROL TAG ExamplesDESCRIPTION <answer/> This answers the call. <hangup/> This informs theserver to hangup the call. <call value=“pstn:14079757500” This elementallows for outbound   maxtime=“15s”/> calls to be created from within avXML script. <conference sessions=”sessionID1, This element allows formultiple sessionID2, sessionID3”/> lines in separate sessions to beconferenced together.

TABLE 6C vXML Media Tags vXML MEDIA TAG Examples DESCRIPTION <play . . ./> <Playaudio> can be used to play an <playnumber format=“say|read”audio file and wait for a terminating value=“12345” digit to be pressed.The element must termdigits=“*#” be located within a larger <block>cleardigits=“TRUE|FALSE”/> structure, which is used to control the<playmoney format=“???” number of repetitions the audio is value=“1.25”played before “timing out.” Like the termdigits=“*#” earlier example of<getDigits>, cleardigits=“TRUE|FALSE”/> <playaudio> can be implemented<playdate format=“ddmmyyhhss” with event handlers to properlyvalue=“1012990732” recognize that the <playaudio> termdigits=“*#”command was halted because a cleardigits=“TRUE|FALSE”/> terminatingdigit was pressed by the user. <playchars format=“?” value=“abcdefgh”termdigits=“*#” cleardigits=“TRUE|FALSE”/> <playtone format=“?”value=“2000hz+1000hz″\” termdigits=“*#” cleardigits=“TRUE|FALSE”/><playaudio format=“audio/msgsm” value=“http://www.blahblah.com/sample.vox” termdigits=“*#” cleardigits=“TRUE|FALSE”/> <recordaudioformat=“audio/msgsm” Like its counterpart <playaudio>, thisvalue=“ftp://www.v.n/msg.wav” element must be contained within atermdigits=“*#” viable <block> structure and utilizecleardigits=“TRUE|FALSE” an event handler such that one maxtime=“30s”generated by a user action to control ptime=“5s”/> it. Its intended useis to leave voicemail messages, record greetings, etc. In the exampleabove, the user would be allowed to speak into the phone to record audiofor up to 30 seconds (with no more than a 5 second pause anywherewithin), and could end the recording segment by pressing either the * or# key. The new audio file would then be saved at www.v.n/msg.wav in theaudio/msgsm format. The clearDigits attribute, again, is used to ensurea “fresh” start during the audio recording, in case one of theterminating digits was pressed prior to initiating the recording.<receivefax format=“audio/tiff-f” Receives a fax.   value=“ftp://w.v.n/msg.tif” maxtime=“5m” ptime=“30s” maxpages=“30”/><sendfax format=“audio/tiff-f” Sends a fax. value=“http://w.v.n/msg.tif”maxtime=“5m” ptime=“30s” maxpages=“30”/> <text format=“?” This is usedto tell the application termdigits=“#” gateway server to use a text-to-cleardigits=“TRUE|FALSE”> speech engine for reading the enclosed text tothe caller. Text to read </text> <vcommand name=“id” This is used totell the application value=“url|vocab-grammar-string”> gateway server touse a speech-to-text engine for voice recognition.

TABLE 6D vXML High Level Tags VXML HIGH LEVEL TAG Examples DESCRIPTION<menu label=“main_menu” Menu is an element that is descended from arepeat=“3” <block> element and a <playAudio> format=“audio/msgsm”element. In this embodiment, an value=“http://w.v.n/msg.wav”<ontermdigit> event handler is used, to cleardigits=“TRUE|FALSE” handlethe event when a terminating digit has termdigits=“567890*#” beenpressed. It is designed to accept a maxtime=“15s” > single digit inputand then check for a   Events matching <onkey> event handler. This      <onkey value=“1”> element is to allow easy-to-use menus, where         </onkey> one key press will move you through an       <onkeyvalue=“2”> application. In the example above (and          </onkey>below), the audio file will be played 3 times     <onmaxtimevalue=“1|2|max”> before “timing out” and moving on in the      </onmaxtime> vXML code    <onhangup>       </onhangup> </menu><inputdigits      label=“input_pin” <InputDigits> is an element that isdescended       repeat=“3” from a <block> element and a <getDigits>      var=“pager_msg” element. It combines the attributes of those      format=“audio/msgsm” two elements into a single element. Like the      value=“http://w.v.n.msg.wav” <menu> element above, <inputDigits>      termdigits=“#*” simplifies the process of making a function to      cleardigits=“TRUE|FALSE” gather digits from the user. It will playa      includetermdigit=“TRUE|FALSE” message (contained in the “value”attribute)       maxdigits=“4” and store the gathered information in the      maxtime=“15s” “var” attribute. In the example, the user has      ptime=“5s”> 15 seconds to enter up to 4 digits (possibly   Eventsfor a PIN code), with a pause of no more      <oninputvalue value=“123”>than 5 seconds between keystrokes. Either        </oninputvalue> the #or * key will end the input process, and       <oninputlength len=“3”>the audio message/prompt will loop 3 times        </oninputlength>before dropping out of the element (i.e.,     <ontermdigit value=“#”>timing out), and proceeding on to the rest of         </ontermdigit> thevXML code.      <onmaxdigits>       </onmaxdigits>      <onmaxtimevalue=“1|2|max”>        </onmaxtime>    <onptime>          </onptime>   <onhangup>         </onhangup> </inputdigits>

EXAMPLES

The following are examples of microXML communication between the sessionmanager 210 and the vXML parser 242.

MicroXML sent from the session manager to the vXML parser Request theparsing of a new file <ACT>  <SID>24601</SID>  <BUF>   <?xmlversion=“1.0” encoding=“UTF-8”?>   <voxeoxml>   <assign var=“rootdir”value=“http://www.voxeo.com/”/>   <block label=“1” repeat=”3“>  <playaudio format=“audio/default” value=“$rootdir;greeting.vox”/>  </block>   </voxeoxml>  </BUF> </ACT> Request the continued parsing ofthe same file <ACT>  <SID>24601</SID> </ACT> Report a basic user eventto the vXML parser <EVT>  <SID>24601</SID>  <ETP>termdigit</ETP> <EVL>#</EVL> </EVT> Report a user event with parameter(s) to the vXMLparser <EVT>  <SID>24601</SID>  <ETP>termdigit</ETP>  <EVL>#</EVL> <P00>assign=varname=value</P00> </EVT>

MicroXML sent from the vXML parser to the session manager Action fromparser <ACT> <TYP>playaudio</TYP><PR1>format=audio/default,value=http://www.voxeo.com/ greeting.vox</PR1></ACT>

The following is an example of a vXML file:

Example vXML file <?xml version=“1.0” encoding=“UTF-8”?> <voxeoxml> <!-- This is a test file to exercise the voxeoXML parser -->  <assignvar=“audiodir” value=“http://www.voxeo.com/audio”/>  <blocklabel=“testlooping” repeat =“3”>  <assign var=“foo” value=“$foo;bar”/> <playaudio   format=“audio/msgsm”   value=“$audiodir;/$foo;.vox”  termdigits=“*”   cleardigits=“true”  /> </block>  <hangup/></voxeoxml>

The example vXML file results in the following corresponding microXMLbeing generated by the vXML parser and sent to the session manager:

The resulting microXML Separate calls to parse are delimited by‘----------------------------’ ---------------------------- <ACT> <SID>24601</SID>  <TYP>cleardigits</TYP> </ACT> <ACT>  <SID>24601</SID> <TYP>playaudio</TYP><PR1>format=audio/msgsm,value=http://www.voxeo.com/audio/bar.vox,termdigits=*</P R1> </ACT> ---------------------------- <ACT> <SID>24601</SID>  <TYP>cleardigits</TYP>  </ACT>  <ACT> <SID>24601</SID>  <TYP>playaudio</TYP><PR1>format=audio/msgsm,value=http://www.voxeo.com/audio/barbar.vox,termdigits=* </PR1> </ACT> ---------------------------- <ACT> <SID>24601</SID>  <TYP>cleardigits</TYP> </ACT> <ACT>  <SID>24601</SID> <TYP>playaudio</TYP><PR1>format=audio/msgsm,value=http://www.voxeo.com/audio/barbarbar.vox,termdigits =*</PR1> </ACT> ----------------------------<ACT>  <SID>24601</SID>  <TYP>hangup</TYP>  <PR1></PR1> </ACT>---------------------------- <ACT>  <SID>24601</SID>  <TYP>EOF</TYP> <PR1></PR1> </ACT>

FIG. 9 is a system block diagram of a network traffic monitoring systemoperating in cooperation with the Distributed Application TelephonyNetwork System of the present invention. The invention contemplates alarger number of enterprises and users will deploy telephonyapplications on the Internet 30 in the form of vXML applications such asvAPP 1, vAPP 2, . . . , vAPP m. These applications are served by anumber of web servers 46 on the Internet. When a call associated withone of the these vAPP enters the Internet, it must be directed to one ofa plurality of application gateway centers, such as vAGC 1, vAGC 2, . .. , vAGC n. In the preferred embodiment, in order to provide the bestquality-of-service, the call is preferably directed to a vAGC having thebest access to the associated vAPP. The invention includes providingmonitoring of the accessibility of the individual vAPPs relative to theplurality of vAGCs. This will enable the call to be directed to the vAGChaving the best access to that associated vAPP.

Each vAGC site is provided with a traffic monitor 400 that periodicallypings the plurality of vAPP sites and detects the return signals. Theresponse time of each vAPP site to any given vAGC is collected by anetwork monitoring server 410. Since each vAPP is associated with a callor dialed number (DN), the network monitoring server computes a listingof DNs versus vAGCs sorted in order of fastest response time. Thisinformation is used to update the DIR0 directory (see FIG. 4A)dynamically. In this way, when a call to a given DN is to be directed toan AGC, a DIR0 lookup will point to the vAGC with the faster access tothe vAPP site associated with the given DN.

While the embodiments of this invention that have been described are thepreferred implementations, those skilled in the art will understand thatvariations thereof may also be possible. Therefore, the invention isentitled to protection within the full scope of the appended claims.

1. A method of processing a telephone call to a call number in anetworked computer telephony system, comprising: providing a pluralityof Extensible Markup Language (XML) documents hosted by a web serveramong a plurality of web servers on the Internet, each of said XMLdocuments constituting a telephony application associated with aspecified call number and including telephony-specific XML scripts withtags instructing how a telephone call to the specified call number is tobe processed; providing a plurality of application gateway centers, eachcapable of processing the telephone call by retrieving and processingthe XML document associated with the call number of the telephone call;periodically pinging from each of the plurality of application gatewaycenters to the plurality of web servers to measure a response time fromeach web servers; and directing the telephone call to be processed bythe application gateway center having a fastest response time from theweb server hosting the XML document associated with the call number. 2.A method as in claim 1, further comprising: providing a directory ofapplication gateway centers in order of faster response time for the webserver hosting the XML document associated with the call number of thetelephone call; and wherein said directing the telephone call furtherincludes: looking up the directory with the telephone call number forthe application gateway center having the fastest response time.
 3. Amethod as in claim 2, further comprising: providing a monitoring serverfor collecting the response time of each web server to any one of theplurality of application gateway centers; dynamically updating thedirectory with a listing of call numbers versus application gatewaycenters, the application gateway centers associated with each callnumber being in order of descending response time.
 4. A networkedcomputer telephony system for processing a telephone call to a callnumber, comprising: a plurality of Extensible Markup Language (XML)documents hosted by a web server among a plurality of web servers on theInternet, each of said XML documents constituting a telephonyapplication associated with a specified call number and includingtelephony-specific XML scripts with tags instructing how a telephonecall to the specified call number is to be processed; a plurality ofapplication gateway centers, each capable of processing the telephonecall by retrieving and processing the XML document associated with thecall number of the telephone call; a plurality of network trafficmonitors, each associated with an individual application gateway centerfor periodically pinging the plurality of web servers to measure aresponse time from each web servers to each application gateway center;and a directory for directing the telephone call to be processed by aapplication gateway center, said directory being dynamically updatedwith a list of application gateway centers in order of fastest responsetime for the web server hosting the XML document associated with thecall number.
 5. A networked computer telephony system for processing atelephone call to a call number, comprising: a plurality of ExtensibleMarkup Language (XML) documents hosted by a web server among a pluralityof web servers on the Internet, each of said XML documents constitutinga telephony application associated with a specified call number andincluding telephony-specific XML scripts with tags instructing how atelephone call to the specified call number is to be processed; aplurality of application gateway centers, each capable of processing thetelephone call by retrieving and processing the XML document associatedwith the call number of the telephone call; a plurality of networktraffic monitors, each associated with an individual application gatewaycenter for periodically pinging the plurality of web servers to measurea response time from each web servers to each application gatewaycenter; a directory for directing the telephone call to be processed bya application gateway center, and means for dynamically updating saiddirectory with a list of application gateway centers in order of fastestresponse time for the web server hosting the XML document associatedwith the call number.